Nice video. was able to set up the same setup without any issues. Thanks!

I feel like it's no exaggeration to say you are a godsend. I've been making my own soft since i was 14ish and since then it's beco my

excellent waiting for other videos for this topic

Thank you for this video. It is really great explanation. 🙏

eventually it all snapped into place and I started learning how to add all the effects, titles, motion text. It was pretty cool to see my

really great video and excellent presentation!

@CodeboardClub

3 жыл бұрын

Thanks

Thank you so much, great tutorial.

TNice tutorials was very helpful thankyou.

Nice video.Thanks!👍

wowo I like this video, I'll try out myself and comment again... Thank you.

thanks you your easy explanation

It works. Thank you!

Thanks, this video is very helpful for beginner. Do you have a guide to dynamic add sip clients and save CDR to MySQL?

hey in this video you used text to speech synthesis to narrate it ,... can you tell me which voice is this and where to get it from ?

this is a great video

Nice video, thanks! What are you using for narration?

Great video, excellent explaination. How do we enable video calls and billing?

Very informative, I am trying to build one that works only on my LAN. How do I modify the sip.conf file because soft phones are failing to register. The Sip Server resides in my Ubuntu PC

Very Nice information, Thanks

@CodeboardClub

2 жыл бұрын

Thanks

Very good video

Thank you for informative video. Is this setup workable for Video Call or different config required, if yes could you pls help setup video call.

Thank you!

Excellent 👌

@CodeboardClub

2 жыл бұрын

Thanks a lot 😊

U earned a follow for this ❤

Great tut. Can you make a video on how to connect to the PSTN, so that one can actually call real-life phones?

@portqasim3063

3 жыл бұрын

is it possible to use mobile device to terminates call via sip server or voipswitch?

really superb thanks

@CodeboardClub

2 жыл бұрын

Thanks

Can you only call numbers on the sip network, or can you call any phone number in the world outside of the sip server e.g. just a random phone number?

Smart video

Thank you.

Great tutorial...thanks

@CodeboardClub

3 жыл бұрын

Thanks

@joeharyar9873

3 жыл бұрын

@@CodeboardClub Hi...from sip phone got video and IM...just to check if video and messaging features also works with the current setup? Any additional setting required? Thanks

Would be interessting to see interfacing with other SIP Gateways, like if it would be possible to have this setup you showed and have the Asterix connected to public phone line (e.g. Telecom of India). So when I call a regular phone number, and the line is avialable to the outside, than i can call through. Kinda using Asterix as Proxy. Calls from the outside go to all internal clients, first to pick up wins (^^).

The best video about voip server config. i have tried to load the link to the sip code for the clients but it does not register on asterisk server. when you run the cmd sip show peers, it says ''no host registered'' Do you know another way to do this?

Thanks!

@Codeboard club Hi! your video was very meaning full. I likes your video. It was very valuable information. but I have question that ( like One office that have 20-30 of total employees and they couple of floor's, everyone have internet connection on their Desk. Now how can they communicate each others via CISCO Or Grandstream IP Phone without Any physical server.) @Codeboard Club I know you are Care's others emotions.

Thanks.

how u change the CLI command to ubuntu@ipaddress? @ 5:45 Also,it showed no devices online after "sip show peers" command? Note: I have done changes in sip.conf (externip)?Pls resolve this

Is it possible to interconnect GSM sim using switch then make international calls but call rate will cost will domestic?

I tried to send HEP of asterisk to Homer, but seems not data capture via Homer. Do you have tutorial how the hep can be send from asterisk to Homer? Thank you

excellent in that we can get the call up and running after watching this 10 min video; rather than 50+ video tutorials from the german technician with aweful english.

@CodeboardClub

3 жыл бұрын

Thanks 😊

@KingKing-xx1jh

3 жыл бұрын

@@CodeboardClub hi can u help me create one pls

@jackodera8893

3 ай бұрын

That's very low of you to speak of someone who has taken time to create useful content. The so called 'german' technician you are referring to takes time to explain each and every action while building a VOIP solution, that is what learning is about... understanding. Not just about having a system work without knowing why you have to make certain decisions, and in the process making it easy for you to troubleshoot. If you build something you are unable to troubleshoot then you end up being a half baked technician creating unsecure systems that can cost an organization a lot of money.

thank you

Thanks for this video

@CodeboardClub

2 жыл бұрын

😊

@jindhu2608

2 жыл бұрын

@@CodeboardClubwaiting for more existing videos bro😊

What if you put this behind load balancer?

Need more video on Asterisk.

its working, i use it for my personal server, 😊

@Asma-cr2jy

Ай бұрын

I am unable to upload sip.conf files in directory as its showing sftp permission denied. Did u get any errors like that?

Great ! Can you make a video on how to connect to windows pc.

Do you have a tutorial on how to do the same in version 20, got an error on trying to load the module?

Sir it is possible to set a system in a way that I am able to receive and make online calls from iphone but the person who I call receives it on a mobile sim .?

Where you got this nice voice to your video? What software you are using to edit it? Your answer would be aprecciated.

Hey man. It's a really nice video. Could you briefly mention the server costs associated with the calls. Like whats would be the cost if we have 1000 users or 10000 users or say a million users.

Hi, this is a great video with excellent explanation. It is possible to have another video for adding number dynamically and allow api to create acc, add number and call and call details ?

@CodeboardClub

3 жыл бұрын

Good idea. I will try to make one video on that.

@hiteshwarjindal421

3 жыл бұрын

@@CodeboardClub i am looking for a custom software for voip and sip would you be looking to work as lead on it?

@smrabbu5592

2 жыл бұрын

@@CodeboardClub hello bro place halp me

Thanks for this video, really helpful, do you have any contacts?

Please make video how to test these calls and take trace on tcp dump and then analysis on wireshark

Question, are you only able to call the numbers that you've set up within the sip server or can you call them from a mobile or landline from any network and if so, will it work vice versa?

@SmileWide1

5 ай бұрын

With VOIP, you can typically call any phone number, regardless of whether it's set up within the SIP server or not. SIP (Session Initiation Protocol) is just one of the protocols used for VOIP communication. If you have a VOIP service configured on your mobile or landline device, you can use it to call any phone number, whether it's on a VOIP network or a traditional landline or mobile network. Similarly, if someone has a VOIP service, you can call their VOIP number from your mobile or landline device. In essence, VOIP allows for interoperability between different networks, so calls can be made and received between VOIP networks and traditional telephone networks.

if you are going to use this then set Tenancy must be "Dedicated" except "shared server" .

hi am unable to connect from linphone to Asterisk voip server on ubuntu 20 on oracle cloud infrastructure.any settings to be made on oracle cloud

Hello, Thanks for this video, works fine. It is possible to make a video call ???

Thumbs Up bro, but show us how to setup and call landline numbers.

how can create client key on Bitvise to connect with EC2?

please help, how to set up the linphone application so that it can be used to send messages/chat? please help i really need it.

Hi, it's really a great video and clear guidelines. I have one issue here. When I call from 7001 to 7002. Call received by 7002. Issue here when I talk from 7001 , 7002 able to listen my voice but when 7002 speaks 7001 not able to listen. How to fix this issue. Fyi I'm using android linphone client in both devices. My Ubuntu server running at Google cloud platforms

@gauravbangre

Жыл бұрын

how did you solve this issue

Can we run this on local LAN instead of public IP

Can you make a video how to use smtp for another computer ..

Video for ASTREKS outbound calling

Can I able to call from my VoIP number to external phone number like jio

Hello, Which software do you use for the voice? So if we need to connect PSTN, we need an asterisk card?

Hey! I'm new to FreePBX. I've set up Raspbx on a Raspberry Pi 4 and would like to make calls using a 4G SIM router (TP Link - TL-MR6500v/ VoLTE, VoIP, VoiceMail). The router works fine on the RJ11 port with a standard phone. I previously attempted to use the chan_dongle module, but it appears that the 4G SIM doesn't work with it. I don't believe the mobile carrier provides any SIP settings for 4G SIMs. Is it possible to connect the VoIP router with Raspbx over Ethernet?

@user-em5wc3zp4v

10 ай бұрын

bro can u suggest any cheap product for 3g calling. my isp don't support volte. i want to make calls on 3g. openwrt + with device can we used?

How can we do voip to local sim call

Also can u elaborate how u created profile 1 in bitvise SSH ? like I'm creating the first time host key ...directions needed and whats the connection between this and sip pem file?

@CodeboardClub

3 жыл бұрын

Just couple of steps. Pls check Google how to create public key profile in bitvise ssh. Thanks.

@gnanaprakaashams4082

3 жыл бұрын

@@CodeboardClub ok thanks

@CodeboardClub

3 жыл бұрын

Hey, did you able to create the key? If not let me know.

@gnanaprakaashams4082

3 жыл бұрын

@@CodeboardClub No...can u help me up?

Can I integrate asteriks with node js

Are you only able to place calls to external numbers? Or is that only for numbers connected to the server?

@gamingfest4411

5 ай бұрын

No

Hi, great explanation. How can we use this for actual phone numbers ?

@morocco70

6 ай бұрын

Hi bro is that video true we can call for free just with internet?

How long did it take for you to compile the aestrix , I used sudo make -j2 and now my cpu is 100% for 1 hour ....and frozen

@CodeboardClub

3 жыл бұрын

You are using AWS free tire Ubuntu? It should take 10 - 15 mins max. Never more than that.

@agentstona

3 жыл бұрын

@@CodeboardClub yeah found out just on small instance 1gb memory not enough needed 2gb

Can anyone let me know, whether this method works with Mac?

How about pure PC to PC. Do you have a video about that? Or I have to install an android emulator on my PC

@emirzmen

2 жыл бұрын

I think you can use microsim. or linphone on PC.

Hello sir Can I connect 8 VOIP phone to one Server (via Ethernet Cable) and use this system locally??

@CodeboardClub

3 жыл бұрын

Yes, you can

@clgproject

3 жыл бұрын

@@CodeboardClub thanks

How do i find aws public ip address, answer plzz.

How video call can be made ?

while overwriting the sip.conf, extensions.conf, voicecall.conf file, i am getting error "Open request has failed with SFTP error PermissionDenied: Permission denied." can anyone help me out

@Asma-cr2jy

Ай бұрын

Having the same error. Did u resolve it?

And how do I add this on my app?

I execute all the commands properly but after I use sudo systemctl start asterisk it shows "system has not been booted with systemd as init system (PID 1). Can't operate. Failed to connect to bus: Host is down" can you please help me with these?

@CodeboardClub

3 жыл бұрын

What server are you using?

@mcmarenjacoba9477

3 жыл бұрын

@@CodeboardClub windows. I’m a totally a newbie please bare with me

I got permission denier error when uploading the configuration files using SFTP.So I have tried "sudo chmod 777 *". Now i am getting another ubuntu error like below: sudo: /etc/sudoers is owned by uid 1000, should be 0 sudo: no valid sudoers sources found, quitting sudo: unable to initialize policy plugin can you help me

@CodeboardClub

3 жыл бұрын

Please send me error screenshot via mail

@netadmin-fraser787

2 жыл бұрын

You need to add the account to the /etc/sudoers file or just use root.

Oh okay ! I've detected your real voice 😀 Which software do you use for the voice??

hello ...anyone thr ...i am having problem with bitvise login

I've tried folowinbg this and was good to the point I tried to load module chan_sip.so at which point I got the error "Error loading module 'chan_sip-so": /usr/lib/asterisk/module/chan_sip.so cannot be open or shared object: No such file or directory.

mine is not working, can someone help me

Hi I mangged to install it and set it up exactly like your tutorial i can call from 7001 to 7002 and from 7002 to 7001 but there is no sound .....

@CodeboardClub

3 жыл бұрын

Ok. Send me all config files via mail

@nikhildusane1129

3 жыл бұрын

@@CodeboardClub Hello I am facing same issue.. I used same config files which you provide

how to call any number

I use PBXnSIP using windows

connectoin failed problem

i need a vidoe on soip with xml call

I manage to have everything running but when I’m uploading configuration file it says permission denied

@CodeboardClub

3 жыл бұрын

Please run "sudo chmod 777 *" in the server (for testing purpose only), to solve permission issue.

@muhammedirfan141

3 жыл бұрын

@@CodeboardClub i got the same issue and tried this solution. but i am getting another error like below sudo: /etc/sudoers is owned by uid 1000, should be 0 sudo: no valid sudoers sources found, quitting sudo: unable to initialize policy plugin

ok can we make free calls to anymobile numbers in India? with this method

@CodeboardClub

3 жыл бұрын

For that you need to pay money to the operator, with which you want to connect (airtel, jio etc.)

And a like 😂

how u got the key pair? 'sip'

@CodeboardClub

3 жыл бұрын

You can select generate key pair, it will ask you to provide a name of keypair and then one pem file will be downloaded.

@gnanaprakaashams4082

3 жыл бұрын

@@CodeboardClub thanks

@gnanaprakaashams4082

3 жыл бұрын

Also can u elaborate how u created profile 1 in bitvise SSH ? like I'm creating the first time host key ...directions needed and whats the connection between this and sip pem file?

PL I love to have my own free internet please can you guide me through to anyone please?

the extension files are failling mate...it's say sftp no have permission to overwrite

@Asma-cr2jy

Ай бұрын

Were u able to resolve it

Türkçe altyazıyı koyan kişinin eline sağlık ö-ö-ö-ö-ö-ö-ö-ö-öptüm bayy, gö-gö-gö-gö-gö-gö-gö-gö-gömmdüm say

Indians' tech are so danger :))

@CodeboardClub

3 жыл бұрын

😊

Ah! Congrats. But I feel stupid. I thought Asterisk would be able to have a phone number so people can call to?

Make video in Call id spoofing